- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] vars scope in dialplan
by Madovsky 07 Jun '10

07 Jun '10

Ok I found the answer... http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg12449.… F ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, June 07, 2010 3:24 PM Subject: vars scope in dialplan example : <extension name="blabla"> <condition> <action application="set" data="test=cake"/> </condition> <condition field="${test}" expression="^(cake)$"> <action application="set" data="prfx=00"/> </condition> <condition field="${sip_to_uri}" expression="^(1\d{10,15})@$${domain}$"> <action application="set" data="getPrfx=${prfx}"/> </condition> </extension> but ${prfx} is empty. I'm almost sure I didn't understand FS vars scope yet.... Thanks Franck

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Gateway Registrations
by Samuel Macedo 07 Jun '10

07 Jun '10

Hi, I want to use the same Gateway but with different users and passwords to make and receive calls. This is the scenario: I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch UserAgents (Sofia External) to this gateway so I can receive inbound calls. And to make a Outbound call I want to decide witch UserAgent I will use. Is there anyway to do this? Regards, -- Samuel Macedo

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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use variables set in sp_profiles/external/gateways
by Madovsky 07 Jun '10

07 Jun '10

Hi, I'm trying to use the varaibles set in external gateway sip profiles in a dialplan with no success. I tried $${vars} and ${var} but the value is empty. Should I use them differently ? Thanks Franck

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mod_conference Wiki - lists
by David Swardstrom 07 Jun '10

07 Jun '10

I have been looking at the mod_conference Wiki page and have checked some of it's information against the current code in mod_conference.c. In particular, I have been interested in the “list” capability which is supported by conference_list(). The following comments will be used to change the “Mod Conference” Wiki page: This applies to the API reference, list entry: <current entry> Output: First line Conference <conference name> (<member_count> member ) Following lines are csv separated list for each conference leg with the following items: id of participiant (starts at 1 after FS restart) Register string of participiant Uuid of participiants call leg Cid number Cid name Status (hear|speak|floor) Volume In Volume Out Energy Level </current entry> <corrected entry> Output: First line Conference <conference name> (<member_count> member[s][ locked]) Following lines are csv separated list for each conference leg with the following items: id of participant (starts at 1 after FS restart) Register string of participant Uuid of participants call leg Caller id number Caller id name Status (hear|speak|talking|video|floor) Volume In Volume Out Energy Level Notes: First line: “locked” reflects the lock/unlock status of the conference. Status: “hear” reflects the mute/unmute status of the member. Status: “speak” reflects the “deaf /undeaf” status of the member. Status: “talking” indicates that the input channel is providing some amount of sound energy. Status: “video” ??? Status: “floor” indicates that this member currently owns the floor </corrected entry> Does anyone have any comments I should add for floor? Is there anything significant to owning the floor? How about video? It seems that video and video conferencing is a capability that is desired and some support is provided. But is it really working? Does anyone use it? Is there an assumption in conferencing that video is currently sourced from only one place? Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

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RTP Proxy
by Gustavo Espeche 07 Jun '10

07 Jun '10

Hi some one know an rtp proxy that work with freeswitch. Thanks. Gustavo Espeche www.easyipcall.com

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FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone?
by Alan Frisch 07 Jun '10

07 Jun '10

Hi, Just tinkering around with FreeSWITCH pretty much using the default configuration, with some customization to connect my Polycom 430 and my ITSP. The problem I am having is that the first few seconds of calls (or early media if available) have no audio. When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP with timestamps, I can see that FreeSwitch is holding onto the 183 and subsequent 200 OK for about 2 seconds before relaying it onto the Polycom. FWIW, server is an Intel Atom 330. Any idea why this is... I am sure it must be a configuration issue somewhere? I did not have this issue with Asterisk, so I am hoping to troubleshoot it and move on with FreeSWITCH Thanks. AF.

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get your job aguinst freeswitch
by Meftah Tayeb 07 Jun '10

07 Jun '10

hello list, i am meftah tayeb, a blind person from algeria that was using freeswitch sunse 1.0.1 release i started firstly with asterisk, that was the very strangett voip application in my life and thank to miconda that redirected me to freeswitch, from asterisk in #openser in 2008 i started learning voip basic in dec 2008 thank to the #freeswitch folk that teached me all this, including firstly anthm, Michael S Collin, brian (BKW), sekil the nice GUI and all other in ogust 2009, algeria telecom and the algeria gouvernmant started blocking sip traffic i was not using it for voip business, but, honestly, just to connect to the public freeswitch conference and the weekely voip users conference i start a complain in my local city, no reply from AT i decided to go to the general office, because i can't use my PC without SIP i got the general directore secrutary ok, he receyved me and heare me saying why you are blocking sip? so he asked me why you need sip? do you do voip business without autorisation? i explaned to him my actual situation and he say: ok, no problem i will open the sip for you, but conditionaly the next mondey you will be here i say ok no problem so, i returned to my home and i see something new: 1. a static ip address linked to my ADSL account 2. sip completly open ok, next monday i was in the general office and i meet the gebneral directotore surprisingly he asked me: do you have a job? i say no but he say yes, you have, tel me i say no aguin and he say you work for algeria telecom ;) he surprised me with this now, he gave me a: good job free house free care with driver so please say thank to the freeswitch project especialy the owner and try to donate to him a much a pocible thank you -- Meftah Tayeb algérie télécom SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz

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Re: [Freeswitch-users] Freeswicth nms
by suvirkumar＠gmail.com 07 Jun '10

07 Jun '10

Budi, Try Nagios for network monitoring Regards Suvir ------Original Message------ From: Budi wibowo Sender: freeswitch-users-bounces(a)lists.freeswitch.org To: freeswitch-users(a)lists.freeswitch.org ReplyTo: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Freeswicth nms Sent: Jun 3, 2010 06:26 Hi all I'm looking for nms that can do minimum alarm,fault, performance and statistic. Alarm could be sent via sms or email. I need the such solution due I have project to involve more than 10 fs box. Both free and commercial solution are ok Regards Budi _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Sent via BlackBerry® from AIS

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Skypopen problem
by bakko 07 Jun '10

07 Jun '10

Hi, I'm trying tu install skypopen on my centOS server witout success. This is my configuration: virtual xen server on linode.com centos 5.5 32 bit freswitch latest trunk xen-devel given from linode If i try to compile the alsa-driver-1.0.20 modifing the files and lines indicates in the wiki when i do make this is the result: /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c: In function 'snd_pcm_oss_open_file' /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: 'fmode_t' undeclared (first use in this function) /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: (Each undeclared identifier is reported only once /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: for each function it appears in.) If i try to compile the alsa-driver-1.0.20 without any modification and then compile dummy.c, when i try to load snd-dummy module this is the result modprobe snd-dummy WARNING: Error inserting snd_page_alloc (/lib/modules/2.6.18.8-linode22/misc/acore/snd-page-alloc.ko): Invalid module format WARNING: Error inserting snd_timer (/lib/modules/2.6.18.8-linode22/misc/acore/snd-timer.ko): Invalid module format WARNING: Error inserting snd_pcm (/lib/modules/2.6.18.8-linode22/misc/acore/snd-pcm.ko): Invalid module format FATAL: Error inserting snd_dummy (/lib/modules/2.6.18.8-linode22/misc/drivers/snd-dummy.ko): Invalid module format modprobe -f snd-dummy WARNING: Error inserting snd_page_alloc (/lib/modules/2.6.18.8-linode22/misc/acore/snd-page-alloc.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.18.8-linode22/misc/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.18.8-linode22/misc/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_dummy (/lib/modules/2.6.18.8-linode22/misc/drivers/snd-dummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) Any idea? BR - Andrea Sannucci -

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